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Enable GN check for webrtc/media

Browse Source 
BUG=webrtc:6828, webrtc:7245
NOTRY=True

Review-Url: https://codereview.webrtc.org/2716143002
Cr-Commit-Position: refs/heads/master@{#17067}
This commit is contained in:
kjellander
2017-03-06 06:04:55 -08:00
committed by Commit bot
parent f949000834
commit a2ef4f94e4
2 changed files with 68 additions and 3 deletions
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1
.gn
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@ -28,6 +28,7 @@ check_targets = [
"//webrtc/common_video/*",
"//webrtc/common_audio/*",
"//webrtc/examples/*",
"//webrtc/media/*",
"//webrtc/modules/audio_coding/*",
"//webrtc/modules/audio_conference_mixer/*",
"//webrtc/modules/audio_device/*",

70
webrtc/media/BUILD.gn
View File

@ -33,6 +33,13 @@ config("rtc_media_warnings_config") {
}
rtc_static_library("rtc_media_base") {
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Enabling GN check triggers cyclic dependency error:
# //webrtc/media:rtc_media_base ->
# //webrtc/pc:rtc_pc ->
# //webrtc/media:media ->
# //webrtc/media:rtc_media_base
check_includes = false
defines = []
libs = []
deps = []
@ -91,12 +98,25 @@ rtc_static_library("rtc_media_base") {
deps += [
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:video_frame_api",
"../api/audio_codecs:audio_codecs_api",
"../base:rtc_base",
"../base:rtc_base_approved",
"../call:call_interfaces",
"../common_video:common_video",
"../p2p",
]
}
rtc_static_library("rtc_media") {
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Enabling GN check triggers cyclic dependency error:
# //webrtc/media:media ->
# //webrtc/media:rtc_media ->
# //webrtc/pc:rtc_pc ->
# //webrtc/media:media
check_includes = false
defines = []
libs = []
deps = []
@ -201,11 +221,25 @@ rtc_static_library("rtc_media") {
"..:webrtc_common",
"../api:call_api",
"../api:transport_api",
"../api:video_frame_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../base:rtc_base",
"../base:rtc_base_approved",
"../call",
"../common_video:common_video",
"../modules/audio_coding:rent_a_codec",
"../modules/audio_device:audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing",
"../modules/video_capture:video_capture_module",
"../modules/video_coding",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../p2p:rtc_p2p",
"../system_wrappers",
"../video",
"../voice_engine",
]
}
@ -225,7 +259,11 @@ if (rtc_include_tests) {
include_dirs = []
public_deps = []
deps = []
deps = [
"../modules/audio_coding:rent_a_codec",
"../modules/audio_processing:audio_processing",
"../p2p:rtc_p2p",
]
sources = [
"base/fakemediaengine.h",
"base/fakenetworkinterface.h",
@ -260,7 +298,17 @@ if (rtc_include_tests) {
}
deps += [
":rtc_media",
":rtc_media_base",
"..:webrtc_common",
"../api:call_api",
"../api:video_frame_api",
"../base:rtc_base",
"../base:rtc_base_approved",
"../base:rtc_base_tests_main",
"../base:rtc_base_tests_utils",
"../call:call_interfaces",
"../test:test_support",
"//testing/gtest",
]
public_deps += [ "//testing/gmock" ]
@ -309,7 +357,10 @@ if (rtc_include_tests) {
testonly = true
defines = []
deps = []
deps = [
"../pc:rtc_pc",
"../test:field_trial",
]
sources = [
"base/codec_unittest.cc",
"base/rtpdataengine_unittest.cc",
@ -380,13 +431,26 @@ if (rtc_include_tests) {
}
deps += [
# TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
":rtc_media",
":rtc_media_base",
":rtc_unittest_main",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../audio",
"../base:rtc_base",
"../base:rtc_base_approved",
"../base:rtc_base_tests_utils",
"../call:call_interfaces",
"../common_video:common_video",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_processing:audio_processing",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_vp8",
"../p2p:rtc_p2p_unittests",
"../system_wrappers:metrics_default",
"../test:test_support",
"../voice_engine:voice_engine",
]
}
}